TITRE Adaptive Packet Video Streaming Over IP Networks - LaBRI

More documents

Recommendations

Info

SIP Terminal MPEG-4 Palm SIP phone Internet Video Conference MPEG-4 Terminal MPEG-4 Server H.323 Cellular phone H.323 Terminal Figure 6-1: Video conferencing between heterogeneous IP terminals The specific motivation underlying this multimedia service is to help define a framework to enable interoperability between heterogynous terminals which use different standard and to allow each technology to talk transparently to others without knowing what utilized protocol in advance for session establishment. A videoconferencing session may consist of multiple video or multiple audio streams, addressing multiple codecs with multi-rate requirements. To permit all connected user to joint conference, one solution consists to use a common protocol for session establishment, which is not obvious. Another solution consists to use new components knows as Media Gateways which essentially allow signaling translation between heterogynous signaling protocols such as SIP, H.323 and DMIF. The term media gateway is not new. It was proposed first for interconnecting telephone circuits and data packets carried over the Internet or over other IP networks. In order to achieve universal IP connectivity and seamless IP multimedia service, the interworking between MPEG-4 and SIP terminals is required. Several points of heterogeneity should be addressed for permitting IP multimedia interworking service: (1) Video and audio formats: digital audio formats will be characterized by many factors such as sampling rates or compression algorithms. Video formats may differ in spatial and temporal resolutions, color depth or compression schemes. This format adaptation issue is addressed by multimedia gateways. (2) Synchronizing and system encoding: each elementary (video, audio, data) stream should be merged to form a single bit-stream for transmission, and they should be synchronized. In Internet, the Real-time Transport Protocol provides temporal and encapsulation functions. (3) Application control: users should be able to control the received stream to simulate interactive VCR-like function (e.g. play, pause, fast rewinding, etc.). IETF Real-Time Streaming Protocol (RTSP) [145], DAVIC-S3 [174] and ISO MPEG DSM-CC [175] are examples of signalling protocols developed for that purpose and require interoperability. (4) Connection control/QoS control: in next generation Internet, QoS could be negotiated by way of different signaling (e.g. RSVP, COPS, MPLS LDP, etc.), while some domains will not provide any QoS guarantee and still perform in best effort, others domains may use different protocols. Thus, there should be a function that translates QoS requests from one domain to another. This can be 140
performed by many ways like SLA (Service Level Agreement) negotiation between two domains using COPS Protocol for example. (5) Call/session control: different IP network domains may adopt different method for reserving resources and maintaining session information. There should be a way of managing two independent sessions to form a composite multimedia session (e.g. a SIP compliant phone call and an MPEG-4 DMIF compliant video call). This later point is addressed in this chapter, in particular it describe service heterogeneity (i.e. call and session control). We depict the design and implementation of an experimental system for IP videoconferencing interworking between ISO MPEG-4 DMIF and IETF SIP signaling protocols. We recall, that this interworking signaling gateway is composed of two core subsystems for supporting two-ways delivery of audio-video streams from a DMIF domain to a SIP domain (i.e. DMIF2SIP subsystem), and from a SIP domain to a DMIF domain (i.e. SIP2DMIF subsystem). The design architecture of the interworking signaling gateway is presented in Section 6.2. 6.1 Related Work 6.1.1 IETF SIP: Session Initiation Protocol Session Initiation Protocol (SIP) [140] is an application-layer control and signaling protocol for creating, modifying and terminating sessions with one or more participants. These sessions include IP multimedia conferences, IP telephone calls and multimedia distribution. SIP has been approved in early 1999 as an official standard by the IETF for signaling communications services on the Internet. SIP can be used to initiate sessions as well as to invite members to sessions. The SIP architecture includes the following protocols: • RTP (Real-Time Transport Protocol) for transporting real time audio, video and data [110], • RTSP (Real-Time Streaming Protocol) for setting up and controlling on-demand media [145], • MGCP (Media Gateway Control Protocol) and Megaco for controlling media gateways [146], • SDP (Session Description Protocol) for describing multimedia sessions [141], • SAP (Session Announcement Protocol) for announcing multicast session [147], • TRIP (Telephony Routing over IP) for locating the best gateway between the Internet and the PSTN (Public Switched Telephone Network) [148], • Suite of resources management and multicast address allocation protocols. 141
Page 1:
UNIVERSITÉ DE VERSAILLES SAINT-QUE
Page 4 and 5:
Résumé Nous constatons aujourd’
Page 6 and 7:
3 Related Work ....................
Page 8 and 9:
5.5.1.2 Simulation Results.........
Page 10 and 11:
List of Figures Figure 1-1: Structu
Page 12 and 13:
Figure 5-15: End-to-end packet tran
Page 14 and 15:
Acknowledgments The first person I
Page 16 and 17:
[11] Toufik Ahmed, Guillaume Burida
Page 18 and 19:
d’accès. Les interactions de QoS
Page 20 and 21:
Métrique MAX_DELAY PREF_MAX_DELAY
Page 22 and 23:
1.1.2 Proposition dun Protocole d'E
Page 24 and 25:
des paquets RTP. Reste toujours le
Page 26 and 27:
Nous pouvons distinguer trois possi
Page 28 and 29:
1.1.2.5.2 Multiplexage des Flux El
Page 30 and 31:
l’horloge est par défaut égale
Page 32 and 33:
La valeur R TCP représente le déb
Page 34 and 35:
Pour ces raisons, nous nous intére
Page 36 and 37:
Telles que illustrés par la Figure
Page 38 and 39:
3. DMIF2SIP vérifie si son propre
Page 40 and 41:
utilisée optionnellement pour loca
Page 42 and 43:
Chapter 2 2 Introduction 2.1 Motiva
Page 44 and 45:
conditions. Based on end-to-end fee
Page 46 and 47:
Chapter 3 3 Related Work Nowadays,
Page 48 and 49:
• multicasting: it takes the stre
Page 50 and 51:
expensive, fixed delivery and recep
Page 52 and 53:
3.1.6 Static vs. Dynamic Channels T
Page 54 and 55:
environment because they require a
Page 56 and 57:
H.261 is called video codec for aud
Page 58 and 59:
VideoScene (VS) VideoObject (VO) Vi
Page 60 and 61:
The emphasis of MPEG-7 will be the
Page 62 and 63:
Temporal scalability involves parti
Page 64 and 65:
multitude of streams. It can switch
Page 66 and 67:
increased linearly in the absence o
Page 68 and 69:
3.2.2.1 End-to-End Retransmission R
Page 70 and 71:
correctly received block. The missi
Page 72 and 73:
epresent user traffic. The delay ca
Page 74 and 75:
3.2.5 IP Signaling Protocols for Pa
Page 76 and 77:
two H.323 endpoints or between an e
Page 79 and 80:
Chapter 4 4 Adaptive Packet Video T
Page 81 and 82:
4.1.1 Video Classification Model Pr
Page 83 and 84:
attributes can refer to the QoS par
Page 85 and 86:
In RBF, 2 D = X − is the distance
Page 87 and 88:
Figure 4-6: Experiment testbed In t
Page 89 and 90:
190 180 170 End-to-end delay MPEG-4
Page 91 and 92:
Sync Layer. The SL-packetized strea
Page 93 and 94:
conveyed by MIME format parameters
Page 95 and 96:
the same value. This timing informa
Page 97 and 98:
4.2.2.1 Fragmentation We propose to
Page 99 and 100:
Timestamp: Indicates the sampling i
Page 101 and 102:
When G is used as generator matrix,
Page 103 and 104:
4.2.3 Performance Evaluation Intens
Page 105 and 106: 0.3 0.25 our proposal classical app
Page 107 and 108: lost packet at the receiver. Conseq
Page 109 and 110: marking scheme. The server must be
Page 111 and 112: The video server adds audio-visual
Page 113 and 114: Marker (TR3CM) [131] to mark the ba
Page 115 and 116: VO1 VO2 AVO Scenario Audio BL EL1 E
Page 117 and 118: Throughput (Kbits) 3000 2500 2000 1
Page 119 and 120: Throughput (Kbits) 3000 2500 2000 1
Page 121 and 122: 4000 3500 3000 2500 2000 1500 1000
Page 123 and 124: 45 40 Original Quality Received Qua
Page 125 and 126: Chapter 5 5 A Dynamic Packet Video
Page 127 and 128: The TCM algorithm is based on a tok
Page 129 and 130: BEGIN Initialize the classification
Page 131 and 132: User Profile (Token Bucket) Traffic
Page 133 and 134: 5.3.3 QoS Management Algorithm Spec
Page 135 and 136: Figure 5-7: Hierarchical aggregatio
Page 137 and 138: • Filters: to put packets into cl
Page 139 and 140: mechanism associated with the Drop
Page 141 and 142: Figure 5-16: Packets loss with IP D
Page 143 and 144: 5.5.2.3 Experimental Results Figure
Page 145 and 146: 1 MPEG-4 Based Layer Stream -AF11 M
Page 147 and 148: 0.2 0.18 MPEG-4 Based Layer Stream
Page 149 and 150: By using our PBNM tool, we allocate
Page 151 and 152: Id Condition Indicator Action 1 Net
Page 153 and 154: The proposed marking algorithm bett
Page 155: Chapter 6 6 A SIP/MPEG-4 Multimedia
Page 159 and 160: 6.1.1.4 SIP Communication Model SIP
Page 161 and 162: • It assign a value (network sess
Page 163 and 164: 6.2.1 SIP2DMIF Subsystem The SIP2DM
Page 165 and 166: Step 1: The MPEG-4 Terminal passes
Page 167 and 168: 1. Set the result C to the empty se
Page 169 and 170: TCP/UDP port, our implementation us
Page 171 and 172: Control & Data Plan Video Rate Cont
Page 173 and 174: characteristics and relevance of ea
Page 175 and 176: References [1] Kazaa Media Desktop
Page 177 and 178: [43] B. Schuster, “Fine granular
Page 179 and 180: [79] D. Bansal and H. Balakrishnan,
Page 181 and 182: [115] N. Laoutaris, I. Stavrakakis,
Page 183 and 184: [154] W. Almesberger, J. H. Salim,
Page 185 and 186: Appendix A A Appendix A: Overview o
Page 187 and 188: Media aware Delivery unaware ISO/IE
Page 189 and 190: A.3.3 Scene Description Streams Sce
Page 191: This appendix presents structure ex
show all

TITRE Adaptive Packet Video Streaming Over IP Networks - LaBRI

Create successful ePaper yourself

Delete template?

Save as template?